Asterisk 13 with FreePBX 13.0.6 Clean install.
Running a Chrome Browser client and a Polycom desk phone on the same LAN as the PBX.
I have 2 extensions. 1 allocated to the Polycom and 1 in use via UCP.
I can ring the Polycom from the UCP and I get a call established with bidirectional audio.
If I ring the UCP extension from the Polycom it goes to voicemail and the logs show the following:-
[2015-10-27 15:52:20] WARNING[63831][C-00000034] app_dial.c: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] netsock2.c: Using SIP VIDEO TOS bits 136
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] netsock2.c: Using SIP VIDEO CoS mark 6
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] netsock2.c: Using SIP RTP TOS bits 184
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] netsock2.c: Using SIP RTP CoS mark 5
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] app_stack.c: SIP/991576-0000005b Internal Gosub(func-apply-sipheaders,s,1) start
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] pbx.c: Executing [s@func-apply-sipheaders:1] NoOp("SIP/991576-0000005b", "Applying SIP Headers to channel") in new stack
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] pbx.c: Executing [s@func-apply-sipheaders:2] Set("SIP/991576-0000005b", "SIPHEADERKEYS=") in new stack
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] pbx.c: Executing [s@func-apply-sipheaders:3] While("SIP/991576-0000005b", "0") in new stack
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] app_while.c: Jumping to priority 7
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] pbx.c: Executing [s@func-apply-sipheaders:8] Return("SIP/991576-0000005b", "") in new stack
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] app_stack.c: Spawn extension (from-internal, 1576, 1) exited non-zero on 'SIP/991576-0000005b'
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] app_stack.c: SIP/991576-0000005b Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] app_dial.c: Called SIP/991576
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] app_dial.c: Connected line update to SIP/1488-0000005a prevented.
[2015-10-27 15:52:20] VERBOSE[63831][C-00000034] app_dial.c: SIP/991576-0000005b is ringing
[2015-10-27 15:52:23] VERBOSE[63831][C-00000034] app_dial.c: Everyone is busy/congested at this time (2:0/0/2)
To make things more confusing... if I take out the Polycom and have a second UCP in it's place... I get the same issue dialling UCP to UCP so it looks like there's an issue ONLY if the call is Inbound to a UCP assigned extension.
Any ideas?
Thanks in advance.
Mike